[SR-Users] Music on hold with freeswitch as SBC

[Gautam Batra]

I'm not able to set up the rtp proxy module. I have entered the following:

loadmodule "rtpproxy.so"
modparam ("rtpproxy", "rtpproxy_sock", "udp:X.Y.Z.W:22222");

Where X.Y.Z.W is the IP address of my machine (same as that of my SIP
server). But the log shows the following errors:

Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR: rtpproxy
[rtpproxy.c:1503]: can't send command to a RTP proxy
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR: rtpproxy
[rtpproxy.c:1538]: proxy <udp:X.Y.Z.W:22222> does not respond, disable it
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING: rtpproxy
[rtpproxy.c:1395]: can't get version of the RTP proxy
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING: rtpproxy
[rtpproxy.c:1432]: support for RTP proxy <udp:X.Y.Z.W:22222> has been
disabled temporarily

Could anyone tell what I'm doing wrong? I tried to run rtpproxy separately
on the given port before starting kamailio (rtpproxy -s udp:X.Y.Z.W:22222),
but it didn't work.



On Wed, Dec 21, 2011 at 2:36 PM, Gautam Batra <gautambatra24 at gmail.com>wrote:

> I am using Freeswitch as an SBC behind Kamailio, and my external calls are
> routed via freeswitch. In those calls the music on hold works as it is
> handled by freeswitch. Ideally I would like to somehow redirect when a call
> is put on hold to the MOH extension. The other option is by using rtpproxy.
> I could not find any documentation on rtpproxy and would really appreciate
> it if someone could lead me to it or give me a brief overview on how to go
> about using rtpproxy_stream2uac to play music whenever a call is put on
> hold.
>
> On Wed, Dec 21, 2011 at 4:50 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>> Hello,
>>
>>
>> On 12/21/11 7:49 AM, Olle E. Johansson wrote:
>>
>>> 20 dec 2011 kl. 22:40 skrev Gautam Batra:
>>>
>>>  Hi,
>>>>
>>>> Thanks for your replies. Is it possible to play an audio file in the
>>>> case of a re-invite directly from kamailio instead of freeswitch by using
>>>> rtpproxy_stream2uac() or something similar?
>>>>
>>> Kamailioi is still a proxy and from the endpoint point of view is not
>>> involved in the media plane. If you managed to do that many
>>> endpoints would ignore the packets or see them as a DOS attack attempt.
>>> Other endpoints might just play them.
>>> In later releases of Asterisk, we lock to the IP address of the peer and
>>> would ignore these. Asterisk used to send music-on-hold
>>> like this before, but we considered it a security issue and started
>>> reinviting to make Asterisk involved in the call again to play
>>> music on hold. Asterisk can do that, because it's a b2bua and is an
>>> endpoint in the call. Kamailio can't initiate a reinvite in the
>>> call.
>>>
>> indeed, kamailio cannot initiate re-invites. You can play an audio file
>> via rtpproxy and rtpproxy_stream2uac() if you use rtpproxy relaying from
>> the beginning of the call. Otherwise, use a sip b2bua which does signaling
>> only until you need to play audio and do re-invites so it gets in media
>> path.
>>
>> Besides Asterisk or FreeSWITCH, a lightweight b2bua that probably offers
>> such functionality is sems (sip express media server) -- I CC-ed Stefan, he
>> can confirm and even give some leads of how to do it.
>>
>> Cheers,
>> Daniel
>>
>>>
>>> /O
>>>
>>>> Gautam
>>>>
>>>> On Mon, Dec 12, 2011 at 4:50 AM, Olle E. Johansson<oej at edvina.net>
>>>>  wrote:
>>>>
>>>> 12 dec 2011 kl. 10:45 skrev Daniel-Constantin Mierla:
>>>>
>>>>  Hello,
>>>>>
>>>>> On 12/9/11 9:04 PM, Gautam Batra wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I have a kamailio sip proxy server with freeswitch acting as SBC. I
>>>>>> want to redirect the call to freeswitch when hold is pressed so that i can
>>>>>> play music on hold. I tried this by using rewritehostport in case of a
>>>>>> re-invite, but the call drops in that case. Could someone please help me
>>>>>> with this?
>>>>>>
>>>>> it is not possible to redirect established calls (it breaks the
>>>>> RFC3261), you have to route the call through freeswitch from its start.
>>>>> Perhaps you can use freeswitch without relaying the media in first place
>>>>> and when you have on hold, you set media patch to go through freeswitch.
>>>>>
>>>> The only solution is having FreeSwitch send an invite with replaces to
>>>> grab the call. The question is how to get it back.
>>>>
>>>> /O
>>>>
>>>>
>>>>  ---
>>> * Olle E Johansson - oej at edvina.net
>>> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
>>>
>>>
>>>
>>>
>>> ______________________________**_________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
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